There is a growing interest in the convergence of the public switched telephone network (PSTN), the Internet and other internets and intranets. The convergence of these networks requires technology that facilitates interworking in a uniform and effective manner. The next generation of unified networks will provide an open and scalable architecture to accommodate multiple vendors and protocols under a common packet network. At the moment, there are several obstacles to providing telephony services on a packet network with the same level of performance and availability as is available on the PSTN today.
The traditional PSTN provides constant bandwidth streams of information between users. These media streams travel over dedicated circuits. On the other hand, packet networks have been prone to packet loss and delays, which affect the quality of streaming media required to carry voice communications. Given the high quality levels associated with the PSTN, subscribers expect and demand traditional quality regardless of the transmission medium.
The bursty nature of packet communications makes controlling communication quality and ensuring sufficient bandwidth very difficult.
The bursty nature of packet communications makes controlling communication quality and ensuring sufficient bandwidth very difficult. Traditional “fat pipe” models, which rely on the overall network to find a way to transmit the packets with the required quality of service levels, have not been completely successful. Further, the integration of call processing for the PSTN and packet networks has proven difficult to manage.
Given the desire to use packet networks as the centerpiece for telephony communications, there is a need for a way to ensure quality of service levels for telephony communication carried at least in part over packet networks and provide call processing in a uniform manner.